| /external/webrtc/talk/media/devices/ |
| macdevicemanagermm.mm | 118 LOG(LS_INFO) << count << " capture device(s) found:";
131 LOG(LS_INFO) << [info UTF8String];
158 LOG(LS_INFO) << [capture_devices count] << " capture device(s) found:";
172 LOG(LS_INFO) << [info UTF8String];
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| yuvframescapturer.cc | 148 LOG(LS_INFO) << "Yuv Frame Generator started";
164 LOG(LS_INFO) << "Yuv Frame Generator stopped";
|
| v4llookup.cc | 74 LOG(LS_INFO) << "Found V4L2 capture device " << device_path;
|
| /external/webrtc/webrtc/base/ |
| testclient_unittest.cc | 67 LOG(LS_INFO) << "Skipping IPv6 test.";
85 LOG(LS_INFO) << "Skipping IPv6 test.";
|
| unittest_main.cc | 96 } else if (rtc::LogMessage::GetLogToDebug() > rtc::LS_INFO) {
97 // Default to LS_INFO, even for release builds to provide better test
99 rtc::LogMessage::LogToDebug(rtc::LS_INFO);
|
| autodetectproxy.cc | 50 LOG(LS_INFO) << "GetProxySettingsForUrl(" << server_url_ << ") - start";
52 LOG(LS_INFO) << "GetProxySettingsForUrl - stop";
59 LOG(LS_INFO) << "AutoDetectProxy found proxy at " << proxy_.address;
61 LOG(LS_INFO) << "AutoDetectProxy initiating proxy classification";
144 LOG(LS_INFO) << "Failed to resolve " << resolver_->address();
213 LoggingSeverity sev = (proxy_.type == PROXY_UNKNOWN) ? LS_ERROR : LS_INFO;
|
| sslstreamadapter_unittest.cc | 69 LOG(LS_INFO) << "Feature disabled... skipping"; \
116 LOG(LS_INFO) << "SSLDummyStreamBase::OnEvent side=" << side_ << " sig="
125 LOG(LS_INFO) << "SSLDummyStreamBase::OnEvent side=" << side_ << " sig="
142 LOG(LS_INFO) << "Closing outbound stream";
317 LOG(LS_INFO) << "SSLStreamAdapterTestBase::OnEvent sig=" << sig;
333 LOG(LS_INFO) << "Setting peer identities by digest";
340 LOG(LS_INFO) << "Setting bogus digest for server cert";
352 LOG(LS_INFO) << "Setting bogus digest for client cert";
410 LOG(LS_INFO) << "Randomly dropping packet, size=" << data_len;
415 LOG(LS_INFO) << "Dropping packet > mtu, size=" << data_len
[all...] |
| proxydetect.cc | 778 LOG(LS_INFO) << "No proxy detected for " << url;
850 LOG(LS_INFO) << "IWMSInternalAdminNetSource::ShutdownProxyContext"
858 LOG(LS_INFO) << "INSNetSourceCreator::Shutdown failed: " << hr;
885 LOG(LS_INFO) << "InternetQueryOption failed: " << GetLastError();
894 LOG(LS_INFO) << "unknown internet access type: "
921 LOG(LS_INFO) << "InternetQueryOption failed: " << GetLastError();
932 LOG(LS_INFO) << "unknown internet access type: " << info->dwAccessType;
[all...] |
| linuxfdwalk_unittest.cc | 56 LOG(LS_INFO) << str.str();
|
| task_unittest.cc | 144 LOG(LS_INFO) << "Task " << stuck_[i].xlat_ << " created with timeout "
187 LOG(LS_INFO) << "Running tasks";
201 LOG(LS_INFO) << "Polling tasks";
213 LOG(LS_INFO) << "Timed out task " << id;
|
| /external/webrtc/talk/media/base/ |
| videoadapter.cc | 190 LOG(LS_INFO) << "VAdapt input interval changed from "
208 LOG(LS_INFO) << "VAdapt Input Resolution Change: "
219 LOG(LS_INFO) << "CPU smoothing is now "
232 LOG(LS_INFO) << "VAdapt output interval changed from "
291 LOG(LS_INFO) << "VAdapt Drop Frame: scaled " << frames_scaled_
314 // for LS_VERBOSE and more for LS_INFO.
330 LOG(LS_INFO) << "VAdapt Frame: scaled " << frames_scaled_
353 LOG(LS_INFO) << "Video Adapter third scaling is now "
429 LOG(LS_INFO) << "VAdapt View Request: "
439 LOG(LS_INFO) << "VAdapt Change Cpu Adapt Min Samples from:
[all...] |
| videocapturer.cc | 183 LOG(LS_INFO) << "Pausing a camera.";
204 LOG(LS_INFO) << "Unpausing a camera.";
232 LOG(LS_INFO) << (muted ? "Muting" : "Unmuting") << " this video capturer.";
271 LOG(LS_INFO) << " Capture Requested " << format.ToString();
279 LOG(LS_INFO) << " Supported " << i->ToString() << " distance " << distance;
295 LOG(LS_INFO) << " Best " << best_format->ToString() << " Interval "
372 LOG(LS_INFO) << "Scaling Screencast from "
430 LOG(LS_INFO) << "Scaling WebCam from "
|
| /external/webrtc/webrtc/p2p/base/ |
| pseudotcp.cc | 195 LOG(LS_INFO) << "Stats[" << buffer << "]";
272 LOG(LS_INFO) << "State: TCP_SYN_SENT";
299 LOG(LS_INFO) << "timeout retransmit (rto: " << m_rx_rto
312 //LOG(LS_INFO) << "m_ssthresh: " << m_ssthresh << " nInFlight: " << nInFlight << " m_mss: " << m_mss;
545 LOG(LS_INFO) << "<-- <CONV=" << m_conv
591 LOG(LS_INFO) << "--> <CONV=" << seg.conv
686 LOG(LS_INFO) << "State: TCP_SYN_RECEIVED";
691 LOG(LS_INFO) << "State: TCP_ESTABLISHED";
730 LOG(LS_INFO) << "rtt: " << rtt
767 LOG(LS_INFO) << "exit recovery"
[all...] |
| p2ptransportchannel.cc | 375 LOG(LS_INFO) << "Set gather_continually to " << gather_continually_;
381 LOG(LS_INFO) << "Set backup connection ping interval to "
394 LOG(LS_INFO) << "Set ICE receiving timeout to " << receiving_timeout_
486 LOG(LS_INFO) << "P2PTransportChannel: " << transport_name() << ", component "
599 LOG(LS_INFO) << "Connection already exists for peer reflexive "
620 LOG(LS_INFO) << "Adding connection from "
659 LOG(LS_INFO) << "Switching best connection on controlled side: "
667 LOG(LS_INFO) << "Not switching the best connection on controlled side yet,"
798 LOG_J(LS_INFO, this) << "Created connection with origin=" << origin << ", ("
[all...] |
| turnport.cc | 302 LOG_J(LS_INFO, this) << "Trying to connect to TURN server via "
406 LOG(LS_INFO) << "TurnPort connected to " << socket->GetRemoteAddress()
426 LOG_J(LS_INFO, this) << "Allocating a new socket after "
617 LOG_J(LS_INFO, this) << "Redirecting from TURN server ["
634 LOG_J(LS_INFO, this) << "Starting TURN host lookup for "
876 LOG_J(LS_INFO, this) << "Scheduled refresh in " << delay << "ms.";
[all...] |
| port.cc | 198 LOG_J(LS_INFO, this) << "Port created";
289 LOG(LS_INFO) << "Received STUN ping "
295 LOG(LS_INFO) << "Received conflicting role from the peer.";
555 LOG_J(LS_INFO, this)
578 // Log at LS_INFO if we send a stun ping response on an unwritable
582 rtc::LS_INFO : rtc::LS_VERBOSE;
619 LOG_J(LS_INFO, this) << "Sending STUN binding error: reason=" << reason
659 LOG_J(LS_INFO, this) << "Port deleted";
795 LOG_J(LS_INFO, this) << "Connection created";
908 // Log at LS_INFO if we receive a ping on an unwritable connection
[all...] |
| /external/webrtc/webrtc/base/java/src/org/webrtc/ |
| Logging.java | 63 LS_SENSITIVE, LS_VERBOSE, LS_INFO, LS_WARNING, LS_ERROR,
118 case LS_INFO:
129 log(Severity.LS_INFO, tag, message);
|
| /external/webrtc/talk/app/webrtc/test/ |
| fakedatachannelprovider.h | 65 LOG(LS_INFO) << "DataChannel connected " << data_channel;
72 LOG(LS_INFO) << "DataChannel disconnected " << data_channel;
|
| /external/webrtc/talk/media/sctp/ |
| sctpdataengine.cc | 134 LOG(LS_INFO) << "SCTP: " << s;
652 LOG(LS_INFO) << debug_name_ << "->SendData(...): EWOULDBLOCK returned";
699 LOG(LS_INFO) << debug_name_ << "->OnInboundPacketFromSctpToChannel(...): "
797 LOG(LS_INFO) << "SCTP_REMOTE_ERROR";
800 LOG(LS_INFO) << "SCTP_SHUTDOWN_EVENT";
803 LOG(LS_INFO) << "SCTP_ADAPTATION_INDICATION";
806 LOG(LS_INFO) << "SCTP_PARTIAL_DELIVERY_EVENT";
809 LOG(LS_INFO) << "SCTP_AUTHENTICATION_EVENT";
817 LOG(LS_INFO) << "SCTP_NOTIFICATIONS_STOPPED_EVENT";
820 LOG(LS_INFO) << "SCTP_SEND_FAILED_EVENT"
[all...] |
| /external/webrtc/talk/media/webrtc/ |
| webrtcvideoengine2.cc | 450 LOG(LS_INFO) << "Creating default receive stream for SSRC=" << ssrc << ".";
477 LOG(LS_INFO) << "WebRtcVideoEngine2::WebRtcVideoEngine2()";
482 LOG(LS_INFO) << "WebRtcVideoEngine2::~WebRtcVideoEngine2";
486 LOG(LS_INFO) << "WebRtcVideoEngine2::Init";
494 LOG(LS_INFO) << "CreateChannel. Options: " << options.ToString();
715 LOG(LS_INFO) << "SetSendParameters: " << params.ToString();
736 LOG(LS_INFO) << "SetRecvParameters: " << params.ToString();
769 LOG(LS_INFO) << "SetRecvCodecs: " << CodecVectorToString(codecs);
790 LOG(LS_INFO)
795 LOG(LS_INFO) << "Changing recv codecs from
[all...] |
| webrtcvideocapturer.cc | 257 LOG(LS_INFO) << "Failed to find best capture format,"
318 LOG(LS_INFO) << "Camera '" << GetId() << "' started with format "
343 LOG(LS_INFO) << "Camera '" << GetId() << "' stopped after capturing "
398 LOG(LS_INFO) << "Capture delay changed to " << delay << " ms";
411 LOG(LS_INFO) << "Captured frame size "
|
| /external/webrtc/talk/app/webrtc/java/jni/ |
| androidmediacodeccommon.h | 50 #define ALOGD LOG_TAG(rtc::LS_INFO, TAG)
|
| /external/webrtc/webrtc/audio/ |
| audio_state.cc | 63 LOG(LS_INFO) << "VoiceEngine error " << err_code << " reported on channel "
|
| /external/webrtc/webrtc/modules/rtp_rtcp/source/ |
| remote_ntp_time_estimator.cc | 70 LOG(LS_INFO) << "RTP timestamp: " << rtp_timestamp
|
| /external/webrtc/talk/session/media/ |
| channel.cc | 193 LOG(LS_INFO) << "Created channel for " << content_name;
218 LOG(LS_INFO) << "Destroyed channel";
269 LOG(LS_INFO) << "Create RTCP TransportChannel for " << content_name()
792 LOG(LS_INFO) << "Channel enabled";
802 LOG(LS_INFO) << "Channel disabled";
822 LOG(LS_INFO) << "Channel writable (" << content_name_ << ")"
830 LOG(LS_INFO) << "Using " << it->local_candidate.ToSensitiveString()
887 LOG(LS_INFO) << "Installing keys from DTLS-SRTP on "
[all...] |
